/* * RTSP muxer * Copyright (c) 2010 Martin Storsjo * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avformat.h" #if HAVE_POLL_H #include <poll.h> #endif #include "network.h" #include "os_support.h" #include "rtsp.h" #include "internal.h" #include "avio_internal.h" #include "libavutil/intreadwrite.h" #include "libavutil/avstring.h" #include "libavutil/time.h" #include "url.h" #define SDP_MAX_SIZE 16384 static const AVClass rtsp_muxer_class = { .class_name = "RTSP muxer", .item_name = av_default_item_name, .option = ff_rtsp_options, .version = LIBAVUTIL_VERSION_INT, }; int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr) { RTSPState *rt = s->priv_data; RTSPMessageHeader reply1, *reply = &reply1; int i; char *sdp; AVFormatContext sdp_ctx, *ctx_array[1]; char url[1024]; if (s->start_time_realtime == 0 || s->start_time_realtime == AV_NOPTS_VALUE) s->start_time_realtime = av_gettime(); /* Announce the stream */ sdp = av_mallocz(SDP_MAX_SIZE); if (!sdp) return AVERROR(ENOMEM); /* We create the SDP based on the RTSP AVFormatContext where we * aren't allowed to change the filename field. (We create the SDP * based on the RTSP context since the contexts for the RTP streams * don't exist yet.) In order to specify a custom URL with the actual * peer IP instead of the originally specified hostname, we create * a temporary copy of the AVFormatContext, where the custom URL is set. * * FIXME: Create the SDP without copying the AVFormatContext. * This either requires setting up the RTP stream AVFormatContexts * already here (complicating things immensely) or getting a more * flexible SDP creation interface. */ sdp_ctx = *s; sdp_ctx.url = url; ff_url_join(url, sizeof(url), "rtsp", NULL, addr, -1, NULL); ctx_array[0] = &sdp_ctx; if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) { av_free(sdp); return AVERROR_INVALIDDATA; } av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri, "Content-Type: application/sdp\r\n", reply, NULL, sdp, strlen(sdp)); av_free(sdp); if (reply->status_code != RTSP_STATUS_OK) return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA); /* Set up the RTSPStreams for each AVStream */ for (i = 0; i < s->nb_streams; i++) { RTSPStream *rtsp_st; rtsp_st = av_mallocz(sizeof(RTSPStream)); if (!rtsp_st) return AVERROR(ENOMEM); dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); rtsp_st->stream_index = i; av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); /* Note, this must match the relative uri set in the sdp content */ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/streamid=%d", i); } return 0; } static int rtsp_write_record(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPMessageHeader reply1, *reply = &reply1; char cmd[1024]; snprintf(cmd, sizeof(cmd), "Range: npt=0.000-\r\n"); ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) return ff_rtsp_averror(reply->status_code, -1); rt->state = RTSP_STATE_STREAMING; return 0; } static int rtsp_write_header(AVFormatContext *s) { int ret; ret = ff_rtsp_connect(s); if (ret) return ret; if (rtsp_write_record(s) < 0) { ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); return AVERROR_INVALIDDATA; } return 0; } int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st) { RTSPState *rt = s->priv_data; AVFormatContext *rtpctx = rtsp_st->transport_priv; uint8_t *buf, *ptr; int size; uint8_t *interleave_header, *interleaved_packet; size = avio_close_dyn_buf(rtpctx->pb, &buf); rtpctx->pb = NULL; ptr = buf; while (size > 4) { uint32_t packet_len = AV_RB32(ptr); int id; /* The interleaving header is exactly 4 bytes, which happens to be * the same size as the packet length header from * ffio_open_dyn_packet_buf. So by writing the interleaving header * over these bytes, we get a consecutive interleaved packet * that can be written in one call. */ interleaved_packet = interleave_header = ptr; ptr += 4; size -= 4; if (packet_len > size || packet_len < 2) break; if (RTP_PT_IS_RTCP(ptr[1])) id = rtsp_st->interleaved_max; /* RTCP */ else id = rtsp_st->interleaved_min; /* RTP */ interleave_header[0] = '$'; interleave_header[1] = id; AV_WB16(interleave_header + 2, packet_len); ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len); ptr += packet_len; size -= packet_len; } av_free(buf); return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); } static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int n; struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0}; AVFormatContext *rtpctx; int ret; while (1) { n = poll(&p, 1, 0); if (n <= 0) break; if (p.revents & POLLIN) { RTSPMessageHeader reply; /* Don't let ff_rtsp_read_reply handle interleaved packets, * since it would block and wait for an RTSP reply on the socket * (which may not be coming any time soon) if it handles * interleaved packets internally. */ ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL); if (ret < 0) return AVERROR(EPIPE); if (ret == 1) ff_rtsp_skip_packet(s); /* XXX: parse message */ if (rt->state != RTSP_STATE_STREAMING) return AVERROR(EPIPE); } } if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams) return AVERROR_INVALIDDATA; rtsp_st = rt->rtsp_streams[pkt->stream_index]; rtpctx = rtsp_st->transport_priv; ret = ff_write_chained(rtpctx, 0, pkt, s, 0); /* ff_write_chained does all the RTP packetization. If using TCP as * transport, rtpctx->pb is only a dyn_packet_buf that queues up the * packets, so we need to send them out on the TCP connection separately. */ if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) ret = ff_rtsp_tcp_write_packet(s, rtsp_st); return ret; } static int rtsp_write_close(AVFormatContext *s) { RTSPState *rt = s->priv_data; // If we want to send RTCP_BYE packets, these are sent by av_write_trailer. // Thus call this on all streams before doing the teardown. This is // done within ff_rtsp_undo_setup. ff_rtsp_undo_setup(s, 1); ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); ff_network_close(); return 0; } AVOutputFormat ff_rtsp_muxer = { .name = "rtsp", .long_name = NULL_IF_CONFIG_SMALL("RTSP output"), .priv_data_size = sizeof(RTSPState), .audio_codec = AV_CODEC_ID_AAC, .video_codec = AV_CODEC_ID_MPEG4, .write_header = rtsp_write_header, .write_packet = rtsp_write_packet, .write_trailer = rtsp_write_close, .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER, .priv_class = &rtsp_muxer_class, };